Market adoption of wireless LAN (WLAN) technology has exploded, as users from a wide range of backgrounds and vertical industries have brought this technology into their homes, offices, and increasingly into the public air space.
This inflection point has highlighted not only the limitations of earlier-generation systems, but the changing role WLAN technology now plays in people's work and lifestyles, across the globe. Indeed, WLANs are rapidly changing from convenience networks to business-critical networks. Increasingly users are depending on WLANs to improve the timeliness and productivity of their communications and applications, and in doing so, require greater visibility, security, management, and performance from their network.
As enterprises and other entities increasingly rely on wireless networks, the capabilities of wireless clients and the uses to which they are put increasingly expand. For example, certain wireless clients, such as laptops and even cell phones with WLAN capabilities, use wireless connections to access the wired computer network and make telephone calls, or engage in other interactive sessions involving multimedia elements, such as voice, video, graphics, and the like. Voice-over-IP (VoIP), for example, describes facilities for managing the delivery of voice information using the Internet Protocol (IP). In general, this means sending voice information in digital form in discrete packets rather than in the traditional circuit-switched protocols of the public switched telephone network. In addition to IP, VoIP uses the real-time protocol (RTP ) to help ensure that packets get delivered in a timely way, and uses the Session Initiation Protocol (SIP) to set up the session implementing the call.
The Session Initiation Protocol (SIP) [IETF Request for Comments [RFC] 2543] is an Internet Engineering Task Force (IETF) standard protocol for initiating an interactive user session that involves multimedia elements such as video, voice, chat, gaming, and virtual reality. Like HTTP or SMTP, SIP works in the Application layer of the Open Systems Interconnection (OSI) communications model. SIP can establish multimedia sessions or Internet telephony calls, and modify, or terminate them. The protocol can also invite participants to unicast or multicast sessions. Because the SIP supports name mapping and redirection services, it makes it possible for users to initiate and receive communications and services from any location, and for networks to identify the users wherever they are.
SIP is based on the request-response paradigm, used to initiate sessions for internet telephony, instant messaging and any other interactive session involving the exchange of data or multimedia elements. To initiate a session, the caller (known as the User Agent Client, or UAC) sends a request (called an INVITE), addressed to the person the caller wants to talk to. In SIP, addresses are URLs. SIP defines a URL format that is very similar to the popular mailto URL. If the user's e-mail address is user@user-domain.com, their SIP URL would be sip:user@user-domain.com. Telephone numbers, mapped to SIP addresses, can also be used. In some systems, this message is not sent directly to the called party, but rather to an entity known as a proxy server. The proxy server is responsible for routing and delivering messages to the called party. The called party then sends a response, accepting or rejecting the invitation, which is forwarded back through the same set of proxies, in reverse order. A proxy can receive a single INVITE request, and send out more than one INVITE request to different addresses. This feature, aptly called “forking,” allows a session initiation attempt to reach multiple locations, in the hopes of finding the desired user at one of them.
The proxy for the called party generally forwards the INVITE to the end system at which the user is currently stationed. SIP REGISTER request and associated functionality provides the proxy an address binding. For example, when a user initiates a SIP client on an end system such as a cell phone, PDA, or laptop, the SIP client registers the binding sip:user@user-domain.com to sip:user@mypda.userpda.com. This allows the proxy to know that the user is actually at mypda, a specific host on the network, connected via a wireless network system. The proxy consults this registration database, and forwards the INVITE to user@mypda.userpda.com. The response is forwarded back through the proxies to the calling user. An acknowledgement is sent.
One problem for VoIP and other sessions requiring real-time or near real time service is the issue of Quality of Service (QoS). The delay in conversations that many VoIP users encounter is caused by the jitter and latency of packet delivery within the Internet itself SIP itself does not allow for reservation of network resources or admission control. Accordingly, SIP relies on other protocols and techniques in order to provide quality of service. To create QoS on the Internet, different classes of service for packets are applied. The IETF has taken two approaches: The first is Integrated Services (RFC 2211 and RFC 2212), also known as INTSERV. The second is Differentiated Services (RFC 2475), or DIFFSERV.
While the prior art addresses QoS for RTP and other real-time traffic over open computer networks, the prior art does not provide a dynamic QoS configuration mechanism for clients in wireless network environments. Accordingly, while QoS may be applied to the packets associated with a call session over a wired computer network, the wireless network systems to which one or both end systems may be connected provide no mechanism for configuring QoS policy for the call session. The prior art also does not provide any mechanism that dynamically responds to situations where the wireless client is handed off to another access point, such as when the user walks into a different coverage area. Embodiments of the present invention substantially fulfill these needs.